Sampling rate conversion apparatus, encoding apparatus decoding apparatus and methods thereof

ABSTRACT

A coding apparatus capable of reducing a circuit scale and also reducing the amount of coding processing calculation is disclosed. In this apparatus, frequency domain conversion section ( 103 ) performs a frequency analysis of the signal sampled at a sampling rate Fx with an analysis length of 2·Na and calculates first spectrum S 1 (k) (0≦k&lt;Na). Band extension section ( 104 ) extends the effective frequency band of first spectrum S 1 (k) to 0≦k&lt;Nb so that a new spectrum can be assigned to the extended area following to the frequency k=Na of first spectrum S 1  (k). Extended spectrum assignment section ( 105 ) assigns extended spectrum S 1 ′(k) (Na≦k&lt;Nb) input to the extended frequency band from outside. Spectral information specification section ( 106 ) outputs information necessary to specify extended spectrum S 1 ′(k) out of the spectrum given from extended spectrum assignment section ( 105 ) as a code.

TECHNICAL FIELD

The present invention relates to a sampling rate conversion apparatus,coding apparatus, decoding apparatus and methods thereof.

BACKGROUND ART

Nowadays, there are many different sampling rates such as 44.1 kHz for acompact disk, 32 kHz or 48 kHz for DAT (Digital Audio Tape), digital VCRor satellite television, 48 kHz or 96 kHz for a DVD audio signal.Therefore, when an internal sampling rate of a decoder of a reproductionapparatus or a recording apparatus is different from the sampling rateof data to be decoded, it is necessary to change the sampling rate. Onesuch conventional apparatus that converts this sampling rate isdescribed, for example, in Patent Document 1.

Also, in recent years, transmission path capacities on a network havebeen significantly improved with the popularity of ADSL (AsymmetricDigital Subscriber Line) and optical fibers in a wired system, practicaluse of W-CDMA (Wideband-Code Division Multiple Access) and wireless LANin a wireless system or the like, and in line with this trend, there aredemands for realization of high sense of realism and high quality byexpanding bandwidth of signal in voice communications.

At present, there are G.726, 729 or the like which are standardized byITU (International Telecommunication Union) as typical schemes forcoding a narrow band signal. Furthermore, examples of typical methodsfor coding a wideband signal include G722, G722.1 of ITU-T(International Telecommunication Union Telecommunication StandardizationSector) and AMR-WB or the like of 3GPP (The 3 rd Generation PartnershipProject).

Moreover, with the intention of being used in various networkenvironments such as an IP (Internet Protocol) network, the voice codingscheme is recently required to realize a scalable function. The scalablefunction means the function capable of decoding a voice signal even frompart of a code. With this scalable function, it is possible to reducethe occurrence frequency of packet loss by decoding a high quality voicesignal using all codes in a communication path under good conditions andtransmitting only part of the code in a communication path under badconditions.

It is also possible to produce effects such as an increase in efficiencyof network resources in multicast communication.

To realize a high quality coding scheme having this scalable function,coding must be performed using signals at various sampling rates. Forexample, if a signal having a sampling rate of 8 kHz is coded using amethod such as G.726, G.729 or the like standardized in ITU-T and itserror signal is further coded in an area of sampling rate of 16 kHz, itis possible to improve quality through an extension of the signalbandwidth and realize scalability.

FIG. 1 is a block diagram showing the typical configuration of a codingapparatus that performs scalable coding. In this example, the number oflayers is N=3 and the sampling rate of a signal layer n is representedFS(n) and suppose FS(1)=16 [kHz], FS(2)=24 [kHz] and FS(3)=32 [kHz].

An acoustic signal (voice signal, audio signal or the like) input todownsampling section 12 through input terminal 11 is downsampled from asampling frequency of 32 kHz to 16 kHz and given to first layer codingsection 13. First layer coding section 13 determines a first code sothat perceptual distortion between the input acoustic signal and thedecoded signal which is generated after the coding becomes a minimum.This first code is sent to multiplexing section 26 and also sent tofirst layer decoding section 14. First layer decoding section 14generates a first layer decoded signal using the first code. Upsamplingsection 15 performs upsampling on the sampling frequency of the firstlayer decoded signal from 16 kHz to 24 kHz and gives the upsampledsignal to subtractor 18 and adder 21.

Furthermore, an acoustic signal input to downsampling section 16 throughinput terminal 11 is downsampled from a sampling frequency of 32 kHz to24 kHz and given to delay section 17. Delay section 17 delays thedownsampled signal by a predetermined duration. Subtractor 18 calculatesthe difference between the output signal of delay section 17 and theoutput signal of upsampling section 15, generates a second layerresidual signal and gives it to second layer coding section 19. Secondlayer coding section 19 performs coding so that the perceptual qualityof the second layer residual signal is improved, determines a secondcode and gives this second code to multiplexing section 26 and secondlayer decoding section 20. Second layer decoding section 20 performsdecoding processing using the second code and generates a second layerdecoded residual signal. Adder 21 calculates the sum between abovedescribed first layer decoded signal and the second layer decodedresidual signal and generates a second layer decoded signal. Upsamplingsection 22 performs upsampling on the sampling frequency of the secondlayer decoded signal from 24 kHz to 32 kHz and gives this signal tosubtractor 24.

Moreover, an acoustic signal input to delay section 23 through inputterminal 11 is delayed by a predetermined duration and given tosubtractor 24. Subtractor 24 calculates the difference between theoutput signal of delay section 23 and the output signal of upsamplingsection 22 and generates a third layer residual signal. This third layerresidual signal is given to third layer coding section 25. Third layercoding section 25 performs coding on the third layer residual signal sothat its perceptual quality is improved, determines a third code andgives the code to multiplexing section 26. Multiplexing section 26multiplexes the codes obtained from first layer coding section 13,second layer coding section 19 and third layer coding section 25 andoutputs the multiplexing result through output terminal 27.

Patent Document 1: Unexamined Japanese Patent Publication No.2000-68948

DISCLOSURE OF THE EMBODIMENT Problems to be Solved by the Invention

However, as mentioned above, the coding apparatus which realizes ascalable function based on a time domain coding scheme such as G.726,729, AMR-WB or the like needs to convert sampling rates of varioussignals (downsampling section 12, upsampling section 15, downsamplingsection 16 and upsampling section 22 in the above described example),which results in a problem that the configuration of the codingapparatus becomes complicated and the amount of coding processingcalculation also increases. Furthermore, the circuit configuration ofthe decoding apparatus that decodes a signal coded by this codingapparatus also becomes complicated and the amount of decoding processingcalculation increases.

It is an object of the present invention to provide a sampling rateconversion apparatus and coding apparatus that can reduce a circuitscale and also reduce the amount of coding processing calculation, adecoding apparatus that decodes a signal coded by this coding apparatusand methods for these apparatuses.

Means for Solving the Problem

The present invention extends an effective frequency band of a spectrumin a frequency domain instead of performing a sampling conversion(especially upsampling) in a time domain and thereby obtains a signalequivalent to a case where a time domain signal is upsampled.

The sampling rate conversion apparatus of the present invention adopts aconfiguration comprising a conversion section that converts an inputtime domain signal to a frequency domain and obtains a first spectrum,an extension section that extends the frequency band of the firstspectrum obtained and an insertion section that inserts a secondspectrum in the extended frequency band of the first spectrum after theextension.

According to this configuration, the input time domain signal isconverted to a frequency domain signal and the frequency band of thespectrum obtained is extended, and it is possible to thereby obtain asignal equivalent to a signal upsampled in the time domain. Furthermore,it is also possible to reduce the circuit scale of the coding apparatusand also reduce the amount of coding processing calculation.

The coding apparatus of the present invention adopts a configurationcomprising a conversion section that performs a frequency analysis of asignal having an input sampling frequency of Fx with an analysis lengthof 2·Na and obtains a first spectrum of an Na point, an extensionsection that extends the frequency band of the first spectrum obtainedto an Nb point and a coding section that specifies a second spectruminserted in the extended frequency band of the first spectrum after theextension and outputs a code representing this second spectrum.

This configuration allows a spectrum having a sampling rate ofFS=Fx·Nb/Na to be obtained without performing any sampling conversion inthe time domain.

In the coding apparatus of the present invention in the above describedconfiguration, the second spectrum is generated based on the firstspectrum.

According to this configuration, it is possible to generate an extendedspectrum based on information obtained by the decoder and therebyrealize a low bit rate.

In the coding apparatus of the present invention in the above describedconfiguration, the second spectrum is determined so as to resemble thespectrum included in a frequency band of Na≦k<Nb out of the spectrumobtained by the frequency analysis of the input signal having a samplingfrequency of Fy at a 2·Nb point.

According to this configuration, it is possible to determine theextended spectrum relative to the spectrum of an original signal andthereby obtain a more accurate extended spectrum.

In the coding apparatus of the present invention in the above describedconfiguration, the coding section divides the frequency band of Na≦k<Nbinto two or more subbands and outputs codes representing the secondspectrum in subband units.

According to this configuration, it is possible to obtain the effect ofgenerating a code having a scalable function.

In the coding apparatus of the present invention in the above describedconfiguration, the signal having a sampling frequency of Fx is a signaldecoded with a lower layer of hierarchical coding.

According to this configuration, the present invention can be applied tohierarchical coding made up of a coding section having a plurality oflayers and the hierarchical coding can be realized only with a minimumsampling conversion.

The decoding apparatus of the present invention adopts a configurationcomprising an acquisition section that performs a frequency analysis ofa signal having a sampling frequency of Fx with an analysis length of2·Na and acquires a first spectrum in a frequency band of 0≦k<Na, adecoding section that receives a code and decodes a second spectrum in afrequency band of Na≦k<Nb, a generation section that combines the firstspectrum and the second spectrum and generates a spectrum in a frequencyband of 0≦k<Nb, and a conversion section that converts the spectrumincluded in the frequency band of 0≦k<Nb to a time domain signal.

According to this configuration, it is possible to decode a codegenerated by the coding apparatus according to any one of the abovedescribed configurations.

In the decoding apparatus of the present invention in the abovedescribed configuration adopts a configuration, the second spectrum isgenerated based on the spectrum in a frequency band of 0≦k<Na.

According to this configuration, it is possible to decode the code usingthe coding method of generating an extended spectrum based oninformation obtained with the decoder and thereby realize a low bitrate.

The decoding apparatus of the present invention in the above describedconfiguration adopts a configuration, further comprising a section thatinserts a specified value into a high-frequency part of the spectrumafter the combination or discards a high-frequency part of the spectrumafter the combination so that the frequency bandwidth of the spectrumafter the combination obtained by the generation section matches apredetermined bandwidth.

According to this configuration, a decoded signal is generated afteradding processing of making the bandwidth of the spectrum constant evenwhen the bandwidth of the spectrum received changes due to factors suchas a condition of a network or the like, and it is possible to therebygenerate a decoded signal at a desired sampling rate stably.

In the decoding apparatus of the present invention in the abovedescribed configuration, the signal having a sampling frequency of Fx isa signal decoded with a lower layer in hierarchical coding.

According to this configuration, it is possible to decode a codeobtained through hierarchical coding made up of the coding sectionhaving a plurality of layers.

ADVANTAGEOUS EFFECT OF THE INVENTION

According to the present invention, it is possible to reduce the circuitscale of the coding apparatus and also reduce the amount of codingprocessing calculation. It is also possible to provide a decodingapparatus that decodes a signal coded by this coding apparatus.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram showing the typical configuration of a codingapparatus that performs scalable coding;

FIG. 2 is a block diagram showing the main configuration of a spectrumcoding apparatus according to Embodiment 1;

FIG. 3 A shows a first spectrum and FIG. 3B shows a spectrum after aneffective frequency band is extended;

FIG. 4A illustrates the effect of processing of extending an effectivefrequency band of a spectrum theoretically;

FIG. 4B illustrates the effect of processing of extending an effectivefrequency band of a spectrum in principle;

FIG. 5 is a block diagram showing the main configuration of a radiotransmission apparatus according to Embodiment 1;

FIG. 6 is a block diagram showing the internal configuration of a codingapparatus according to Embodiment 1;

FIG. 7 is a block diagram showing the internal configuration of aspectrum coding section according to Embodiment 1;

FIG. 8 is a block diagram showing a variation of the spectrum codingsection according to Embodiment 1;

FIG. 9 is a block diagram showing the main configuration of a radioreception apparatus according to Embodiment 1;

FIG. 10 is a block diagram showing the internal configuration of adecoding apparatus according to Embodiment 1;

FIG. 11 is a block diagram showing the internal configuration of aspectrum decoding section according to Embodiment 1;

FIG. 12A and FIG. 12B illustrate the processing carried out by a bandextension section according to Embodiment 1;

FIG. 13 illustrates how a spectrum is processed at a combining sectionand a time domain conversion section according to Embodiment 1 togenerate a decoded signal;

FIG. 14A is a block diagram showing the main configuration on thetransmitting side when the coding apparatus according to Embodiment 1 isapplied to a wired communications system;

FIG. 14B is a block diagram showing the main configuration on thereceiving side when the decoding apparatus according to Embodiment 1 isapplied to a wired communications system;

FIG. 15 is a block diagram showing the main configuration of a decodingapparatus according to Embodiment 2;

FIG. 16 is a block diagram showing the internal configuration of aspectrum decoding section according to Embodiment 2;

FIG. 17 illustrates processing of a correction section according toEmbodiment 2 in more detail;

FIG. 18 illustrates processing of the correction section according toEmbodiment 2 in more detail;

FIG. 19 further illustrates the operation of the spectrum decodingsection according to Embodiment 2;

FIG. 20A further illustrates the operation of the spectrum decodingsection according to Embodiment 2;

FIG. 20B further illustrates the operation of the spectrum decodingsection according to Embodiment 2;

FIG. 21 shows the main configuration of a communications systemaccording to Embodiment 3; and

FIG. 22 shows the main configuration of a communications systemaccording to Embodiment 4.

BEST MODE FOR CARRYING OUT THE INVENTION

Now, embodiments of the present invention will be described in detailwith reference to the accompanying drawings.

Embodiment 1

FIG. 2 is a block diagram showing the main configuration of spectrumcoding apparatus 100 according to Embodiment 1 of the present invention.

Spectrum coding apparatus 100 according to this embodiment is providedwith sampling rate conversion section 101, input terminal 102, spectralinformation specification section 106 and output terminal 107.Furthermore, sampling rate conversion section 101 has frequency domainconversion section 103, band extension section 104 and extended spectrumassignment section 105.

A signal sampled at a sampling rate Fx is input to spectrum codingapparatus 100 through input terminal 102.

Frequency domain conversion section 103 converts a time domain signal toa frequency domain signal (frequency domain conversion) by performing afrequency analysis of this signal with an analysis length of 2·Na andcalculates first spectrum S1(k)(0≦k<Na). Then, first spectrum S1(k)calculated is given to band extension section 104. Here, a modifieddiscrete cosine transform (MDCT) is used for the frequency analysis. TheMDCT is characterized in that an analysis frame and a successive frameare overlapped by half on top one another and analysis is performed, andthereby distortion between the frames is canceled using an orthogonalbasis whereby the first half portion of the analysis frame becomes anodd function and the second half portion of the analysis frame becomesan even function. As the method of the frequency analysis, it is alsopossible to use a discrete Fourier transform (DFT), discrete cosinetransform (DCT) or the like.

Band extension section 104 allocates a new area (frequency band) so thata new spectrum can be assigned to the extended area following to thefrequency k=Na of input first spectrum S1(k) and extends the effectivefrequency band of first spectrum S1(k) to 0≦k<Nb. The processing ofextending this effective frequency band will be explained in detaillater.

Extended spectrum assignment section 105 assigns extended spectrumS1′(k)(Na≦k<Nb) input from outside to the frequency band extended byband extension section 104 and outputs it to spectral informationspecification section 106.

Spectral information specification section 106 outputs informationnecessary to specify extended spectrum S1′(k) out of the spectrum givenfrom extended spectrum assignment section 105 as the code through outputterminal 107. This code is information which shows the subband energy ofextended spectrum S1(k) and information which shows an effectivefrequency band or the like. Details thereof will also be describedlater.

Next, details of the processing carried out by above described bandextension section 104 to extend the effective frequency band of firstspectrum S1(k) will be explained using FIG. 3A and FIG. 3B.

FIG. 3A shows first spectrum S1(k) given from frequency domainconversion section 103 and FIG. 3B shows spectrum S1(k) after aneffective frequency band is extended by band extension section 104. Bandextension section 104 allocates the area in which new spectralinformation can be inserted in the frequency band where frequency k offirst spectrum S1(k) is shown in the range of Na≦k<Nb. The size of thisnew area is expressed by “Nb−Na”.

Here, Nb is determined from the relationship between sampling rate Fx ofthe signal given from outside through input terminal 102, analysislength 2·Na in frequency domain conversion section 103 and sampling rateFy of the signal decoded by a decoding section (not shown). Morespecifically, Nb is set by the following expression: $\begin{matrix}{{Nb} = {{Na} \cdot \frac{Fy}{Fx}}} & \left( {{Expression}\quad 1} \right)\end{matrix}$

Furthermore, sampling rate Fy of the signal decoded by the decodingsection when Nb has been determined is determined by the followingexpression: $\begin{matrix}{{Fy} = {{Fx} \cdot \frac{Nb}{Na}}} & \left( {{Expression}\quad 2} \right)\end{matrix}$

For example, when the coding section is designed under a condition ofNa=128, Fx=16 kHz and a decoded signal of Fy=32 kHz is generated by thedecoding section, it is necessary to set Nb=128·32/16=256. Therefore, inthis case, an area of 128≦k<256 is allocated. Furthermore, as anotherexample, when the coding section is designed under a condition ofNa=128, Nb=384, Fx=8 kHz, the sampling rate of the decoded signalgenerated by the decoding section becomes Fy=8·384/128=24 kHz.

FIG. 4A and FIG. 4B illustrate the effect of the processing of extendingthe effective frequency band of the spectrum carried out by bandextension section 104 in principal. FIG. 4A shows the spectrum Sa(k)obtained when performing a frequency analysis of the signal of samplingrate Fx with an analysis length of 2·Na. The horizontal axis shows afrequency and the vertical axis shows spectrum intensity.

The signal effective frequency band is 0 to Fx/2 from the Nyquisttheorem. The analysis length is 2·Na at this time, and therefore, therange of frequency index k is 0≦k<Na and the frequency resolution ofspectrum Sa(k) is Fx/(2·Na). On the other hand, when spectrum Sb(k)obtained by the frequency analysis with an analysis length of 2·Nb afterthe same signal is upsampled to sampling rate Fy is shown in FIG. 4B,the signal effective frequency band is extended to 0 to Fy/2 and therange of frequency index k is 0≦k<Nb. Here, when Nb satisfies(Expression 1), frequency resolution Fy/(2·Nb) of spectrum Sb(k) isequal to Fx/(2·Na). That is, spectrum Sa(k) in band 0≦k<Na is equal tospectrum Sb(k). Looking from the opposite point of view, this means thatwhen the band of spectrum Sa(k)(0≦k<Na) is extended to Nb, spectrumSb(k) matches the spectrum obtained by the frequency analysis with theanalysis length of 2·Nb after upsampling the signal of sampling Fx tosampling Fy. Using this principle, it is possible to obtain a spectrumequivalent to the upsampled signal without upsampling in the timedomain.

In this way, sampling rate conversion section 101 converts the inputtime domain signal to a frequency domain signal and extends theeffective frequency band of the spectrum obtained, and therefore, it ispossible to obtain a spectrum equivalent to the spectrum obtained byconverting the frequency of the signal upsampled in the time domain.

Since the signal output from sampling rate conversion section 101 is asignal in the frequency domain, when the signal in the time domain isnecessary, it may be possible to provide a time domain conversionsection and perform reconversion to the time domain. In above describedexample, sampling rate conversion section 101 is set inside spectrumcoding apparatus 100, and therefore the signal is input to spectralinformation specification section 106 as the same frequency domainsignal without being returned to the time domain signal and a code isgenerated.

Here, the coding rate of the code output from spectral informationspecification section 106 changes by adjusting the selection of theextended spectrum input to extended spectrum assignment section 105 andthe specific method of the spectral information by spectral informationspecification section 106. That is, the processing of part in samplingrate conversion section 101 has a large influence on the coding, too.This means that spectrum coding apparatus 100 realizes the conversion ofthe sampling rate and coding of the input signal at the same time.

Here, for simplicity of explanation, the case where an extended spectrumis assigned to the original spectrum by extended spectrum assignmentsection 105 has been explained as an example, but the processing carriedout by spectral information specification section 106 is intended tooutput the information necessary to specify an extended spectrum as thecode, and it is sufficient that at least the extended spectrum to beassigned is specified, and therefore the extended spectrum need notalways be actually assigned.

Furthermore, upsampling has been explained here as an example of thesampling rate conversion but the above described principle can also beapplied to downsampling.

FIG. 5 is a block diagram showing the main configuration of radiotransmission apparatus 130 when coding apparatus 120 according to thisembodiment is mounted on the transmitting side of the radiocommunications system.

This radio transmission apparatus 130 includes coding apparatus 120,input apparatus 131, A/D conversion apparatus 132, RF modulationapparatus 133 and antenna 134.

Input apparatus 131 converts sound wave W11 audible to human ears to ananalog signal which is an electric signal and outputs it to A/Dconversion apparatus 132. A/D conversion apparatus 132 converts thisanalog signal to a digital signal and outputs it to coding apparatus 120(signal S1). Coding apparatus 120 encodes input digital signal S1,generates a coded signal and outputs it to RF modulation apparatus 133(signal S2). RF modulation apparatus 133 modulates coded signal S2,generates a modulated coded signal and outputs it to antenna 134.Antenna 134 transmits the modulated coded signal as radio wave W12.

FIG. 6 is a block diagram showing the internal configuration of abovedescribed coding apparatus 120. Here, the case where hierarchical coding(scalable coding) is performed will be explained as an example.

Coding apparatus 120 includes input terminal 121, downsampling section122, first layer coding section 123, first layer decoding section 124,delay section 126, spectrum coding section 100 a, multiplexing section127 and output terminal 128.

Acoustic signal S1 of sampling rate Fy is input to input terminal 121.Downsampling section 122 applies downsampling to signal S1 input throughinput terminal 121 and generates and outputs a signal having a samplingrate Fx. First layer coding section 123 encodes this downsampled signaland outputs the code obtained to multiplexing section (multiplexer) 127and also outputs it to first layer decoding section 124. First layerdecoding section 124 generates a decoded signal of the first layer basedon this code.

On the other hand, delay section 126 gives a delay of a predeterminedlength to signal S1 input through input terminal 121. Suppose themagnitude of this delay has the same value as a time delay generatedwhen the signal has passed through downsampling section 122, first layercoding section 123 and first layer decoding section 124. Spectrum codingsection 100 a performs spectrum coding using signal S3 having a samplingrate Fx output from first layer decoding section 124 and signal S4having a sampling rate Fy output from delay section 126 and outputsgenerated code S5 to multiplexing section 127. Multiplexing section 127multiplexes the code obtained by first layer coding section 123 withcode S5 obtained by spectrum coding section 100 a and outputs themultiplexed signal as output code S2 through output terminal 128. Thisoutput code S2 is given to RF modulation apparatus 133.

FIG. 7 is a block diagram showing the internal configuration of abovedescribed spectrum coding section 100 a. This spectrum coding section100 a has a basic configuration similar to that of spectrum codingapparatus 100 shown in FIG. 2, and therefore the same components areassigned the same reference numerals and explanations thereof will beomitted.

A feature of spectrum coding section 100 a is to give extended spectrumS1′(k)(Na≦k<Nb) using the spectrum of input signal S3 having samplingrate Fy. According to this, since a target signal to determine extendedspectrum S1′(k) is given, and therefore the accuracy of extendedspectrum S1′(k) improves and as a result, the effect of leading toquality improvement is obtained.

Frequency domain conversion section 112 performs a frequency analysis ofsignal S4 of the sampling rate Fy input through input terminal 111 withanalysis length 2·Nb and obtains second spectrum S2(k)(0≦k<Nb). Here,suppose that the relationship shown in (Expression 1) holds betweensampling frequencies Fx, Fy and analysis lengths Na, Nb.

Spectral information specification section 106 determines the code whichshows extended spectrum S1′(k). Here, extended spectrum S1′(k) isdetermined using second spectrum S2(k) obtained by frequency domainconversion section 112. Spectral information specification section 106determines a code in two steps; a step of determining the shape ofextended spectrum S1′(k) and a step of determining the gain of extendedspectrum S1′(k).

The step of determining the shape of extended spectrum S1′(k) will beexplained below first.

In this step, extended spectrum S1′(k) is determined using the band0≦k<Na of first spectrum S1(k). As the specific method thereof, firstspectrum S1(k) which is separated by a certain fixed value C on thefrequency axis as shown in the following expression is copied toextended spectrum S1′(k).S1′(k)=S1(k−C)(Na≦k<Nb)   (Expression 3)

Here, C is a predetermined fixed value and needs to satisfy thecondition of C≦Na. According to this method, the information indicatingthe shape of extended spectrum S1′(k) is not output as the code.

As another method, instead of above described fixed value C, it may bealso possible to use variable T which takes a value in a certainpredetermined range T_(MIN) to T_(MAX) and output value T′ of variable Twhen the shape of extended spectrum S1′(k) is most similar to that ofsecond spectrum S2(k) as part of the code. At this time, extendedspectrum S1′(k) is shown by the following expression:S1′(k)=S1(k−T′)(Na≦k<Nb)   (Expression 4)

Next, the step of determining the gain of extended spectrum S1′(k)obtained by spectrum information specification section 106 will beexplained below.

The gain of extended spectrum S1′(k) is determined so as to match thepower in the band Na≦k<Nb of second spectrum S2(k). More specifically,according to the following expression, deviation V of the power iscalculated, and an index obtained by quantizing this value is output asthe code through output terminal 107. $\begin{matrix}{V = \sqrt{\frac{\sum\limits_{k = {Na}}^{{Nb} - 1}{S\quad 2(k)^{2}}}{\sum\limits_{k = {Na}}^{{Nb} - 1}{S\quad 1^{\prime}(k)^{2}}}}} & \left( {{Expression}\quad 5} \right)\end{matrix}$

Furthermore, it may be also possible to adopt a mode in which extendedspectrum S1′(k) is divided into a plurality of subbands and determine acode independently for each subband. In such a case, in the step ofdetermining the shape of extended spectrum S1′(k), it is possible todetermine T′ expressed by (Expression 4) for each subband and output itas the code and determine only one common T′ and output it as the code.Then, in the step of determining the gain of extended spectrum S1′(k),deviation V(j) of the power is calculated for each subband and an indexobtained by quantizing this value is output as the code through outputterminal 107. The amount of variation of the power for each subband isexpressed by the following expression: $\begin{matrix}{{V(j)} = \sqrt{\frac{\sum\limits_{k = {{BL}{(j)}}}^{{BH}{(j)}}{S\quad 2(k)^{2}}}{\sum\limits_{k = {{BL}{(j)}}}^{{BH}{(j)}}{S\quad 1^{\prime}(k)^{2}}}}} & \left( {{Expression}\quad 6} \right)\end{matrix}$where, j denotes a subband number and BL(j) denotes a frequency indexcorresponding to the minimum frequency of the jth subband, BH(j) denotesa frequency index corresponding to the maximum frequency of the jthsubband. By adopting the configuration in which a code is output foreach subband in this way, it is possible to realize the scalablefunction.

Apart from the mode in which second spectrum S2(k) is calculated asshown in FIG. 7, it is also possible to adopt a mode (spectrum codingsection 100 b) in which the signal of sampling rate Fy is LPC-analyzedas shown in FIG. 8. That is, it is also possible to LPC-analyze thesignal of sampling rate Fy, obtain an LPC coefficient and determineextended spectrum S1′(k) using this LPC coefficient. In thisconfiguration, it is possible to apply a DFT to the LPC coefficient andconvert it to spectral information and determine extended spectrumS1′(k) using this spectrum.

In this way, according to the coding apparatus of this Embodiment, it ispossible to reduce the circuit scale of the coding apparatus and alsoreduce the amount of coding processing calculation.

In addition to the above described effect, the following effect isobtained when the coding apparatus of this Embodiment is applied toscalable coding.

As in the case of the conventional art, when the sampling rate isconverted in the time domain, the input signal needs to be passedthrough a low pass filter (hereinafter referred to as “LPF”) to avoidaliasing. Generally, when filtering processing is performed in the timedomain, a time delay occurs in the output signal with respect to theinput signal. When an FIR-type filter is applied to the LPF, the filterorder must be increased to make its cutoff characteristic steep, whichproduces not only a substantial increase of the amount of calculationbut also a time delay equivalent to the half of sample numbers of thefilter order.

For example, when a 256th-order filter is applied to a signal having asampling frequency Fs=24 kHz, a delay equal to or greater than 5 ms isproduced by only a sampling rate conversion. The occurrence of such adelay, when the 256th-order filter is applied to a bidirectional speechcommunication, causes a problem because the reaction of the other sideof communication is perceived as if it becomes slower.

Furthermore, when using an IIR-type filter for the LPF, the cutoffcharacteristic can be made steeper even if the order is reducedcomparatively and the delay never becomes as big as that of the FIR-typefilter. However, in the case of using the IIR-type filter, it is notpossible to design such a filter that the amount of delay which occursin all the frequencies like the FIR-type filter becomes constant. Inscalable coding, when a signal after the sampling rate conversion issubtracted from the input signal during the scalable coding, it isnecessary to give a predetermined delay amount to the input signalaccording to the time delay of the signal after the sampling rateconversion. However, when an IIR-type LPF is used, the amount of delaywith respect to the frequency is not constant, and therefore the problemthat the subtraction processing cannot be performed accurately occurs.

The coding apparatus of this embodiment can solve these problems whichoccur during scalable coding.

FIG. 9 is a block diagram showing the main configuration of radioreception apparatus 180 which receives a signal transmitted from radiotransmission apparatus 130.

This radio reception apparatus 180 is provided with antenna 181, RFdemodulation apparatus 182, decoding apparatus 170, D/A conversionapparatus 183 and output apparatus 184.

Antenna 181 receives a digital coded acoustic signal as radio wave W12,generates a digital received coded acoustic signal which is an electricsignal and gives it to RF demodulation apparatus 182. RF demodulationapparatus 182 demodulates the received coded acoustic signal fromantenna 181, generates a demodulated coded acoustic signal S11 and givesit to decoding apparatus 170.

Decoding apparatus 170 receives digital demodulated coded acousticsignal S11 from RF demodulation apparatus 182, performs decodingprocessing, generates digital decoded acoustic signal S12 and gives itto D/A conversion apparatus 183. D/A conversion apparatus 183 convertsdigital decoded acoustic signal S12 from decoding apparatus 170,generates an analog decoded voice signal and gives it to outputapparatus 184. Output apparatus 184 converts the analog decoded voicesignal which is an electric signal to vibration of the air and outputsit as sound wave W13 audible to human ears.

FIG. 10 is a block diagram showing the internal configuration of abovedescribed decoding apparatus 170. Also here, a case where a signalgenerated by hierarchical coding is decoded will be explained as anexample.

This decoding apparatus 170 is provided with input terminal 171,separation section 172, first layer decoding section 173, spectrumdecoding section 150 and output terminal 176.

Code S11 generated by hierarchical coding is input from RF demodulationapparatus 182 to input terminal 171. Separation section 172 separatesdemodulated coded acoustic signal S11 input through input terminal 171and generates a code for first layer decoding section 173 and a code forspectrum decoding section 150. First layer decoding section 173 decodesthe decoded signal of sampling rate Fx using the code obtained fromseparation section 172 and gives this decoded signal S13 to spectrumdecoding section 150. Spectrum decoding section 150 performs spectrumdecoding which will be described later on code S14 separated byseparation section 172 and signal S13 of sampling rate Fx generated byfirst layer decoding section 173, generates decoded signal S12 ofsampling rate Fy and outputs this through output terminal 176.

FIG. 11 is a block diagram showing the internal configuration of abovedescribed spectrum decoding section 150.

This spectrum decoding section 150 includes input terminals 152, 153,frequency domain conversion section 154, band extension section 155,decoding section 156, combining section 157, time domain conversionsection 158 and output terminal 159.

Signal S13 sampled at sampling rate Fx is input to input terminal 152.Furthermore, code S14 related to extended spectrum S1′(k) is input toinput terminal 153.

Frequency domain conversion section 154 performs a frequency analysis oftime domain signal S13 input from input terminal 152 with an analysislength of 2·Na and calculates first spectrum S1(k). A modified discretecosine transform (MDCT) is used as the frequency analysis method. TheMDCT is characterized in that an analysis frame and a successive frameare overlapped by half on top one another and analysis is performed, andthereby distortion between the frames is canceled using an orthogonalbasis whereby the first half portion of the analysis frame becomes anodd function and the second half portion of the analysis frame becomesan even function. First spectrum S1(k) obtained in this way is given toband extension section 155. As the frequency analysis method, a discreteFourier transform (DFT), discrete cosine transform (DCT) or the like canalso be used.

Band extension section 155 allocates an area so that a new spectrum canbe assigned to the extended area following to the frequency k=Na ofinput first spectrum S1(k) and ensures that the band of first spectrumS1(k) become 0≦k<Nb. First spectrum S1(k) whose band has been extendedis output to combining section 157.

On the other hand, decoding section 156 decodes code S14 related toextended spectrum S1′(k) input through input terminal 153, obtainsextended spectrum S1′(k) and outputs it to combining section 157.

Combining section 157 combines first spectrum S1(k) given from bandextension section 155 and extended spectrum S1′(k). This combination isrealized by inserting extended spectrum S1′(k) in the band Na≦k<Nb offirst spectrum S1(k). First spectrum S1(k) obtained through thisprocessing is output to time domain conversion section 158.

Time domain conversion section 158 applies time domain conversionprocessing which is equivalent to the inverse conversion of thefrequency domain conversion carried out by spectrum coding section 100 aand generates signal S12 in the time domain through a multiplication ofan appropriate window function and a overlap-add processing. Signal S12in the time domain generated in this way is output as the decoded signalthrough output terminal 159.

Next, the processing to be carried out by band extension section 155will be explained using FIG. 12A and FIG. 12B.

FIG. 12A shows first spectrum S1(k) given from frequency domainconversion section 154. FIG. 12B shows the spectrum obtained as a resultof the processing of band extension section 155 and an area in which newspectral information can be stored is allocated in the band in whichfrequency k is expressed in the range of Na≦k<Nb. The size of this newarea is expressed by Nb−Na. Nb depends on the relationship amongsampling rate Fx of the signal given from input terminal 152, analysislength 2·Na of frequency domain conversion section 154 and sampling rateFy of the signal decoded by spectrum decoding section 150, and it ispossible to set Nb according to the following expression:$\begin{matrix}{{Nb} = {{Na} \cdot \frac{Fy}{Fx}}} & \left( {{Expression}\quad 7} \right)\end{matrix}$

Also, when Nb is determined, sampling rate Fy of the signal decoded byspectrum decoding section 150 is determined by the following expression:$\begin{matrix}{{Fy} = {{Fx} \cdot \frac{Nb}{Na}}} & \left( {{Expression}\quad 8} \right)\end{matrix}$

For example, when a decoded signal having a sampling rate of Fy=32 kHzis generated by spectrum decoding section 150 under the condition wherethe sampling rate of the input signal is Fx=16 kHz and the analysislength of frequency domain conversion section 154 is Na=128, it isnecessary to set Nb=128·32/16=256 at band extension section 155.Therefore, in this case, band extension section 155 allocates the areaof 128≦k<256. In another example, when the sampling rate of the inputsignal is Fx=8 kHz, the analysis length of frequency domain conversionsection 154 is Na=128 and the amount of extension of band extensionsection 155 is Nb=384, the sampling rate of the decoded signal generatedat spectrum decoding section 150 is Fy=8·384/128=24 kHz.

FIG. 13 shows how a decoded signal is generated through the processingof combining section 157 and time domain conversion section 158.

Combining section 157 inserts extended spectrum S1′(k)(Na≦k<Nb) in theband of Na≦k<Nb of first spectrum S1(k) where a band has been extendedand sends combined first spectrum S1(k)(0≦k<Nb) obtained by insertion totime domain conversion section 158. Time domain conversion section 158generates a decoded signal in the time domain and this allows a decodedsignal having a sampling rate of FS (=Fx·Nb/Na).

In this way, the decoding apparatus according to this embodiment candecode a signal coded by the coding apparatus according to thisembodiment.

Here, the case where the coding apparatus or the decoding apparatusaccording to this embodiment is applied to a radio communications systemhas been explained as an example, but the coding apparatus or thedecoding apparatus according to this embodiment can also be applied to awired communications system as shown below.

FIG. 14A is a block diagram showing the main configuration of thetransmitting side when the coding apparatus according to this embodimentis applied to a wired communications system. The same components asthose shown in FIG. 5 are assigned the same reference numerals andexplanations thereof will be omitted.

Wired transmission apparatus 140 includes coding apparatus 120, inputapparatus 131 and A/D conversion apparatus 132 and the output thereof isconnected to network N1.

The input terminal of A/D conversion apparatus 132 is connected to theoutput terminal of input apparatus 131. The input terminal of codingapparatus 120 is connected to the output terminal of A/D conversionapparatus 132. The output terminal of coding apparatus 120 is connectedto network N1.

Input apparatus 131 converts sound wave W11 audible to human ears to ananalog signal which is an electric signal and gives it to A/D conversionapparatus 132. A/D conversion apparatus 132 converts an analog signal toa digital signal and gives it to coding apparatus 120. Coding apparatus120 encodes an input digital signal, generates a code and outputs it tonetwork N1.

FIG. 14B is a block diagram showing the main configuration of thereceiving side when the decoding apparatus according to this embodimentis applied to a wired communications system. The same components asthose shown in FIG. 9 are assigned the same reference numerals andexplanations thereof will be omitted.

Wired reception apparatus 190 includes reception apparatus 191 connectedto network N1, decoding apparatus 170, D/A conversion apparatus 183 andoutput apparatus 184.

The input terminal of reception apparatus 191 is connected to networkN1. The input terminal of decoding apparatus 170 is connected to theoutput terminal of reception apparatus 191. The input terminal of D/Aconversion apparatus 183 is connected to the output terminal of decodingapparatus 170. The input terminal of output apparatus 184 is connectedto the output terminal of D/A conversion apparatus 183.

Reception apparatus 191 receives a digital coded acoustic signal fromnetwork N1, generates a digital received acoustic signal and gives it todecoding apparatus 170. Decoding apparatus 170 receives the receivedacoustic signal from reception apparatus 191, carries out decodingprocessing on this received acoustic signal, generates a digital decodedacoustic signal and gives it to D/A conversion apparatus 183. D/Aconversion apparatus 183 converts the digital decoded voice signal fromdecoding apparatus 170, generates an analog decoded voice signal andgives it to output apparatus 184. Output apparatus 184 converts theanalog decoded acoustic signal which is an electric signal to vibrationof the air and outputs it as sound wave W13 audible to human ears.

In this way, according to the above described configuration, it ispossible to provide a wired transmission/reception apparatus havingoperations and effects similar to those of the above describedtransmission/reception apparatus.

Embodiment 2

FIG. 15 is a block diagram showing the main configuration of decodingapparatus 270 according to Embodiment 2 of the present invention. Thisdecoding apparatus 270 has a basic configuration similar to that ofdecoding apparatus 170 shown in FIG. 10, and therefore the samecomponents are assigned the same reference numerals and explanationsthereof will be omitted.

A feature of this embodiment is to generate a decoded signal having adesired sampling rate by correcting maximum frequency index Nb of firstspectrum S1(k)(0≦k<Nb) after combination processing to desired value Nc.

Spectrum decoding section 250 carries out spectrum decoding using codeS14 separated by separation section 172, signal S13 of sampling rate Fxgenerated by first layer decoding section 173 and coefficient Nc (signalS21) input through input terminal 271. Spectrum decoding section 250then outputs the decoded signal of sampling rate Fy obtained throughoutput terminal 176. When the analysis length of frequency domainconversion of spectrum decoding section 250 is 2·Na, sampling rate Fy ofthe decoded signal is expressed Fy=Fx·Nc/Na.

FIG. 16 is a block diagram showing the internal configuration of abovedescribed spectrum decoding section 250.

Coefficient Nc input through input terminal 271 is given to correctionsection 251 and time domain conversion section 158 a.

Correction section 251 corrects the effective band of first spectrumS1(k)(0≦k<Nb) given from combining section 157 to 0≦k<Nc based oncoefficient Nc (signal S21) given through input terminal 271. Correctionsection 251 then gives first spectrum S1(k)(0≦k<Nc) after the bandcorrection to time domain conversion section 158 a.

Time domain conversion section 158 a applies conversion processing tofirst spectrum S1(k)(0≦k<Nc) given from correction section 251 under ananalysis length of 2·Nc according to coefficient Nc given through inputterminal 271, performs a multiplication with an appropriate windowfunction and a overlap-add processing, generates a signal in the timedomain and outputs it through output terminal 159. The sampling rate ofthis decoded signal becomes FS=Fx·Nc/Na.

FIG. 17 and FIG. 18 are diagram illustrating processing by correctionsection 251 in more detail.

FIG. 17 shows processing by correction section 251 when Nc<Nb. The bandof first spectrum S1(k) (signal S21) given from combining section 157 is0≦k<Nb. Therefore, correction section 251 deletes a spectrum in therange of Nc≦k<Nb so that the band of this first spectrum S1(k) becomes0≦k<Nc. As a result, first spectrum S1(k)(0≦k<Nc) (signal S22) obtainedis given to time domain conversion section 158 a and decoded signal S23in the time domain is generated. The sampling rate of this decodedsignal S23 becomes FS=Fx·Nc/Na.

FIG. 18 also shows processing by correction section 251, but in thiscase Nc>Nb. The band of first spectrum S1(k) (signal S25) given fromcombining section 251 is 0≦k<Nb as in the case of FIG. 17. Correctionsection 251 extends the band of Nb≦k<Nc so that the band of this firstspectrum S1(k) becomes 0≦k<Nc and assigns a specific value (e.g. zero)to the area. As a result, first spectrum S1(k) (0≦k<Nc) (signal S26) isgiven to time domain conversion section 158 a and decoded signal S27 inthe time domain is generated. The sampling rate of this decoded signalS27 becomes FS=Fx·Nc/Na.

The operation of spectrum decoding section 250 will be further explainedusing FIG. 19, FIG. 20A and FIG. 20B.

First, suppose that the code input through input terminal 153 changesfrom one frame to another. That is, suppose that there are three bandsin the band from combining section 157 as shown in FIG. 19; 0≦k<Na (bandR1), 0≦k<Nb1 (band R2), 0≦k<Nb2 (band R3) (note that Na<Nb1<Nb2) and oneof these bands is selected for each frame.

FIG. 20A illustrates the operation of the spectrum decoding section 250when coefficient Nc is equal to Nb2, and FIG. 20B illustrates theoperation of spectrum decoding section 250 when coefficient Nc is equalto Nb1.

These figures express that the band of the spectrum obtained in the i-thframe is any one of R1, R2, R3. Furthermore, processing 1 shows theprocessing of inserting a zero value in the band of Nb1≦k<Nb2,processing 2 shows the processing of inserting a zero value in the bandof Na≦k<Nb2, processing 3 shows the processing of deleting the band ofNb1≦k<Nb2 and processing 4 shows the processing of inserting a zerovalue in the band of Na≦k<Nb1.

First, the case of FIG. 20A will be explained.

In this figure, in the 0th frame to the 1st frame and the 7th frame tothe 8th frame, since the band of the spectrum is R3, that is, the bandof first spectrum S1(k) is 0≦k<Nb2, and therefore correction section 251outputs first spectrum S1(k)(0≦k<Nb2) to time domain conversion section158 a without applying any processing.

Furthermore, in the 2nd frame to the 4th frame and the 9th frame, sincethe band of the spectrum is R2, that is, the band of first spectrumS1(k) is 0≦k<Nb1, correction section 251 extends the band of firstspectrum S1(k) to Nb2, inserts a zero value in the band of Nb1≦k<Nb2 andthen outputs first spectrum S1(k)(0≦k<Nb2) to time domain conversionsection 158 a.

On the other hand, the band of the spectrum is R1 in the 5th frame tothe 6th frame, that is, the band of first spectrum S1(k) is 0≦k<Na, andtherefore correction section 251 extends the band of first spectrumS1(k) to Nb2, inserts a zero value in the range of Na≦k<Nb2 and thenoutputs first spectrum S1(k)(0≦k<Nb2) to time domain conversion section158 a.

Next, the case of FIG. 20B will be explained.

In this figure, in the 2nd frame to the 4th frame and the 9th frame, theband of the spectrum is R2, that is, the band of first spectrum S1(k) is0≦k<Nb1, and therefore correction section 251 outputs first spectrumS1(k)(0≦k<Nb1) to time domain conversion section 158 a without applyingany processing.

Furthermore, in the 0th frame to the 1st frame, and the 7th frame to the8th frame, the band of the spectrum is R3, that is, the band of firstspectrum S1(k) is 0≦k<Nb2, correction section 251 deletes the band ofNb1≦k<Nb2, and then outputs first spectrum S1(k)(0≦k<Nb1) to time domainconversion section 158 a.

On the other hand, in the 5th frame to the 6th frame, the band of thespectrum is R1, that is, the band of first spectrum S1(k) is 0≦k<Na, andtherefore correction section 251 extends the band of first spectrumS1(k) to Nb1, inserts a zero value in the band of Na≦k<Nb1, and thenoutputs first spectrum S1(k)(0≦k<Nb1) to time domain conversion section158 a.

According to the this embodiment, even when the effective frequency bandof received first spectrum S1(k) changes temporally, appropriatecoefficient Nc is given in this way, and it is possible to therebyobtain a decoded signal at a desired sampling rate stably.

Embodiment 3

FIG. 21 shows the main configuration of a communications systemaccording to of Embodiment 3 of the present invention.

A feature of this embodiment is to deal with a case where the effectivefrequency band of first spectrum S1(k) received on the receiving sidechanges temporally depending on the condition of the communicationnetwork (communication environment).

Hierarchical coding section 301 applies the hierarchical codingprocessing shown in Embodiment 1 to the input signal of sampling rate Fyand generates a scalable code. Here, suppose the generated code is madeup of information (R31) on band 0≦k<Ne, information (R32) on bandNe≦k<Nf and information (R33) on band Nf≦k<Ng. Hierarchical codingsection 301 gives this code to network control section 302.

Network control section 302 transfers a code given to from hierarchicalcoding section 301 to hierarchical decoding section 303. Here, networkcontrol section 302 discards part of the code to be transferred tohierarchical decoding section 303 according to the condition of thenetwork. For this reason, the code to be input to hierarchical decodingsection 303 is any one of the code made up of information R31 to R33when there is no code to be discarded, the code made up of informationR31 and R32 when the code of information R33 is discarded and the codemade up of information R31 when the code of information R32 and R33 isdiscarded.

Hierarchical decoding section 303 applies the hierarchical decodingmethod shown in Embodiment 1 or Embodiment 2 to a given code andgenerates a decoded signal. When Embodiment 1 is applied to hierarchicaldecoding section 303, sampling rate Fz of the output decoded signalbecomes Fy (because Fz=Fy·Ng/Ng). Furthermore, when Embodiment 2 isapplied to hierarchical decoding section 303, it is possible to set thesampling rate of the decoded signal according to desired coefficient Nc,and sampling rate Fz of the decoded signal becomes Fy·Nc/Ng.

In this way, according to the this embodiment, even when the effectivefrequency band of first spectrum S1(k) received on the receiving sidechanges temporally depending on the condition of the communicationnetwork, the receiving side can obtain the decoded signal of a desiredsampling rate stably.

Embodiment 4

FIG. 22 shows the main configuration of a communications systemaccording to Embodiment 4 of the present invention.

A feature of this embodiment is that even when one code generated by onehierarchical coding section is simultaneously transmitted to pluralhierarchical decoding sections having different decodable sampling rates(different decoding capacities), the receiving side can handle the codeand obtain decoded signals having different sampling rates.

Hierarchical coding section 401 applies the coding processing shown inEmbodiment 1 to the input signal of sampling rate Fy and generates ascalable code. Here, suppose the generated code is made up ofinformation (R41) on band 0≦k<Nh, information (R42) on band Nh≦k<Ni andinformation (R43) on band Ni≦k<Nj. Hierarchical coding section 401 givesthis code to first hierarchical decoding section 402-1, secondhierarchical decoding section 402-2 and third hierarchical decodingsection 402-3 respectively.

First hierarchical decoding section 402-1, second hierarchical decodingsection 402-2 and third hierarchical decoding section 402-3 apply thehierarchical decoding method shown in Embodiment 1 or Embodiment 2 to agiven code and generate a decoded signal. First hierarchical decodingsection 402-1 performs decoding processing when coefficient Nc=Nj,second hierarchical decoding section 402-2 performs decoding processingof when coefficient Nc=Ni and third hierarchical decoding section 402-3performs decoding processing of when coefficient Nc=Nh.

First hierarchical decoding section 402-1 performs decoding processingof when coefficient Nc=Nj and generates a decoded signal. Sampling rateF1 of this decoded signal becomes Fy (because F1=F·Nj/Nj).

Second hierarchical decoding section 402-2 performs decoding processingof when coefficient Nc=Ni and generates a decoded signal. Sampling rateF2 of this decoded signal becomes Fy Ni/Nj.

Third hierarchical decoding section 402-3 performs decoding processingof when coefficient Nc=Nh and generates a decoded signal. Sampling rateF3 of this decoded signal becomes Fy·Nh/Nj.

In this way, according to this embodiment, the transmitting side cantransmit a code without considering the decoding capacity on thereceiving side, and therefore it is possible to suppress the load of acommunication network. Furthermore, decoded signals having plural typesof sampling rates can be generated in a simple configuration and with asmaller amount of calculation.

The coding apparatus or the decoding apparatus according to the presentinvention can also be mounted on a communication terminal apparatus anda base station apparatus in a mobile communications system, and it ispossible to thereby provide a communication terminal apparatus and abase station apparatus having operations and effects similar to thosedescribed above.

Here, the case where the present invention is constructed by hardwarehas been explained as an example but the present invention can also berealized by software.

The present application is based on Japanese Patent Application No.2003-341717 filed on Sep. 30, 2003, entire content of which is expresslyincorporated by reference herein.

INDUSTRIAL APPLICABILITY

The coding apparatus and the decoding apparatus according to the presentinvention have the effect of realizing scalable coding in a simpleconfiguration and with a small amount of calculation and are suitablefor use in a communications system such as an IP network.

1-17. (canceled)
 18. A sampling rate conversion apparatus comprising: aconversion section that obtains a spectrum from a time domain signalhaving an arbitrary sampling rate through a frequency domain conversion;and a determining section that determines a bandwidth of an extendedspectrum which is added to said spectrum and extends the bandwidth ofsaid spectrum based on said arbitrary sampling rate and desired outputsampling rate.
 19. The sampling rate conversion apparatus according toclaim 18, wherein said spectrum whose bandwidth is extended isequivalent to a signal obtained by upsampling the time domain signalhaving said arbitrary sampling rate up to said desired output samplingrate.
 20. A coding apparatus comprising: a conversion section thatobtains a spectrum from a time domain signal having an arbitrarysampling rate through a frequency domain conversion; a determiningsection that determines the bandwidth of an extended spectrum which isadded to said spectrum and extends the bandwidth of said spectrum basedon said arbitrary sampling rate and desired output sampling rate; ageneration section that generates said extended spectrum based on saidspectrum; and a coding section that encodes said spectrum and saidextended spectrum.
 21. The coding apparatus according to claim 20,wherein said generation section generates said extended spectrum similarto said spectrum based on said spectrum.
 22. The coding apparatusaccording to claim 20, wherein said coding section divides said extendedspectrum into two or more subbands and performs coding in subband units.23. A scalable coding apparatus comprising: a first coding section thatencodes a first band of a voice signal or audio signal; and a secondcoding section that encodes a second band of said voice signal or saidaudio signal, wherein said second coding section comprises: a conversionsection that obtains a spectrum from a time domain signal having a firstsampling rate obtained by said first coding section through a frequencydomain conversion; a determining section that determines a bandwidth ofan extended spectrum which is added to said spectrum and extends thebandwidth of said spectrum based on said first sampling rate and thesecond sampling rate which is equivalent to said second band; ageneration section that generates said extended spectrum based on saidspectrum; and a coding section that encodes said spectrum and saidextended spectrum.
 24. A communication terminal apparatus comprising thecoding apparatus according to claim
 20. 25. A base station apparatuscomprising the coding apparatus according to claim
 20. 26. A decodingapparatus comprising: an acquisition section that acquires codinginformation generated by a coding apparatus; a first conversion sectionthat obtains a spectrum from a time domain signal having an arbitrarysampling rate included in said coding information through a frequencydomain conversion; a determining section that determines a bandwidth ofan extended spectrum which is added to said spectrum and extends thebandwidth of said spectrum based on the sampling rate of said specifictime domain signal and a desired output sampling rate; a generationsection that generates said extended spectrum based on said codinginformation; and a second conversion section that obtains a time domainsignal from said spectrum and said extended spectrum through a timedomain conversion.
 27. The decoding apparatus according to claim 26,wherein said generation section generates said extended spectrum similarto said spectrum based on said coding information.
 28. The decodingapparatus according to claim 26, wherein said extended spectrum isdivided into two or more subbands and includes coding information ofsaid extended spectrum which is coded in subband units.
 29. A scalabledecoding apparatus comprising: a first decoding section that decodes afirst band of a voice signal or audio signal; and a second decodingsection that decodes said second band of said voice signal or said audiosignal, wherein said second decoding section comprises: a firstconversion section that obtains a spectrum from a time domain signal ofa first sampling rate obtained by said first decoding section through afrequency domain conversion; a determining section that determines abandwidth of an extended spectrum which is added to said spectrum andextends the bandwidth of said spectrum based on said first sampling rateand a second sampling rate which is equivalent to said second band; ageneration section that generates said extended spectrum based on codinginformation generated by a scalable coding apparatus; and a secondconversion section that obtains a time domain signal from said spectrumand said extended spectrum through a time domain conversion.
 30. Thescalable decoding apparatus according to claim 29, further comprising athird decoding section that decodes a third band of said voice signal orsaid audio signal, wherein said third decoding section generates aspectrum from a time domain signal of said first sampling rate, appliesprocessing such as zero insertion or deletion to the high frequency partof the spectrum, obtains a spectrum of said third band and converts thespectrum of said third band to a time domain signal.
 31. A communicationterminal apparatus comprising the decoding apparatus according to claim26.
 32. A base station apparatus comprising the decoding apparatusaccording to claim
 26. 33. A sampling rate conversion method comprising:a step of obtaining a spectrum from a time domain signal having anarbitrary sampling rate through a frequency domain conversion; and astep of determining the bandwidth of an extended spectrum which is addedto said spectrum and extends the bandwidth of said spectrum based onsaid arbitrary sampling rate and desired output sampling rate.
 34. Acoding method comprising: a step of obtaining a spectrum from a timedomain signal having an arbitrary sampling rate through a frequencydomain conversion; a step of determining a bandwidth of an extendedspectrum which is added to said spectrum and extends the bandwidth ofsaid spectrum based on said arbitrary sampling rate and desired outputsampling rate; a step of generating said extended spectrum based on saidspectrum; and a step of coding said spectrum and said extended spectrum.35. A decoding method comprising: a step of acquiring coding informationgenerated by a coding apparatus; a step of obtaining a spectrum from atime domain signal having an arbitrary sampling rate included in saidcoding information through a frequency domain conversion; a step ofdetermining a bandwidth of an extended spectrum which is added to saidspectrum and extends the bandwidth of said spectrum based on thesampling rate of said specific time domain signal and desired outputsampling rate; a step of generating said extended spectrum based on saidcoding information; and a step of obtaining a time domain signal fromsaid spectrum and said extended spectrum through a time domainconversion.